Understanding Video Codecs

Codecs are one of the last things I wrapped my head around when I got into video; they’re not exactly an engaging subject, and all the technical terms you run into can be overwhelming. But as a Sony shooter it quickly became apparent that I needed to dive in and figure this out before it became a real limiting factor. I should preface this by saying that overall, I’m really happy with my a7S II’s. The full-frame sensor, lens system, dynamic range, and overall image quality have me sold. But no system’s perfect, and the weak point of Sony cameras has always been the codec. So let’s figure out how to work with it.

To put it simply, a codec is a file format. There’s a DVD codec, a Blu-Ray codec, a YouTube codec; all the media you watch has been encoded into a format that fits the playback system. Consumer codecs are fairly standardized, but when you’re on the professional side, there are a huge range of options to choose from (hence the confusion). Luckily, despite the vast number of individual codecs out there, the principles are pretty simple. Frame rate (frames per second) and resolution (number of pixels) are easy to understand, and both are set by the camera before you press record. The main factors controlled by the codec are bit rate and bit depth.
Bit Rate

Bit rate is a measure of how much total information (e.g. visual detail) can be captured. Think of it this way; there’s a huge amount of information passing from the lens into the camera, but a consumer camera can’t process all of it at once. Because of this, it has to compromise and throw out some of that information. What the codec’s trying to do is figure out which parts are important, and which parts can be discarded. For example; in a shot of a skyscraper against a blue sky, the codec will try to preserve the fine details of the building while assigning less data to the solid backdrop of the sky.

Now, when you’re using high bit rates (say, 400 megabits per second and up), there’s a lot less need for compromise; the details can be preserved across the whole image. But when you’ve only got a limited amount of data to work with, you have to compromise more. And that’s where we run into trouble; the Sony a7S II has a maximum bit rate of 100 megabits per second, which is a far cry from current professional formats (DNxHR HQX is about 800 megabits per second). As such, it’s throwing away a lot of data, and it doesn’t always make the right call. For example, here’s a wide shot from a recent live recording (click here for full size):


As you can see, the codec is preserving the detail in the back curtain and the drum set, but it’s throwing away a lot of detail in the faces. Of course, that’s the opposite of what we want! A colleague of mine ran into an extreme example of this while shooting a concert with an LCD screen behind the artists; the codec put all the information into the background while turning the performer’s faces into a blurry mess. So yes, this is a real problem.

One workaround is to shoot with a shallow depth of field. When you keep the subject in sharp focus while blurring the background, you’re essentially forcing the codec to keep the detail where it belongs. As an example, here’s another shot from the same show (click here for full size):


An image like this is ideal content for a limited codec. But there is no in-camera fix if your shot just has a lot of detail in it. For that, we’ll have to turn to an external solution.
External Recorders

There’s a reason external recorders are so popular right now; they allow you to send your video signal to dedicated hardware that can use a much higher bit rate. Here’s another shot from the show, this one recorded to an Atomos recorder (click here for full size):


Despite the fact that there’s a lot more going on in the background, the detail in the face and hair is preserved; and that’s with a bit rate of only 200 megabits per second. (Note that this image was not denoised, so there’ll be some digital artifacts visible.) By increasing the bit rate, we vastly improve the detail and quality of the footage we can capture. However, bit rate is just one side of the equation.
Bit Depth

Where bit rate is the amount of information in video, bit depth is the kind of information that can be recorded; more specifically, the kind of color. 8-bit video is capable of storing about 17 million colors, where 10-bit video is capable of storing over 1 billion. Think of it this way; let’s say you have a gradient that’s going from one color to the next. With 10-bit color, you have about 60 times as many “steps” in between compared to 8-bit. Here’s what that looks like in practice (courtesy of Dave Dugdale):


As you can see, the colors on the left transition sharply around the nose and mouth, creating some chromatic errors (looks like pink noise here). On the right, the skin tones blend much more smoothly. This becomes even more obvious when you’re grading footage, as pushing the exposure or changing the color exacerbates those errors.

Problem is, the Sony a7S II doesn’t output 10-bit footage. And if it doesn’t output it, an external recorder can’t capture it. So there’s no way around this; we’re stuck with sharper transitions between colors and the artifacts that result.

While using an external recorder can get you a much more detailed image, unfortunately the a7S II will always be limited by the bit depth of its codec. And that’s why I’m hoping the a7S III will have 10-bit output. If so, we’ll have a camera that’s capable of producing very high-quality footage that’ll hold up against cameras many times its price. Regardless, I’d always recommend using an external recorder with any a7 series camera, and setting your exposure and white balance as accurately as possible to cut down on color correction in post. And lastly, I’d highly recommend denoising all your footage before grading; that’ll be the subject of my next post.


Resolve 14 vs. Premiere Pro CC


I’ve been using Adobe Premiere since I first started editing nearly ten years ago, and while it’s been gaining momentum as a professional editor, I’d always been curious about Blackmagic’s DaVinci Resolve. With version 14 hot off the presses, I thought I’d dive in, give it a spin and see if it could replace Premiere as my go-to NLE. After 12 hours (straight), here are my thoughts:



Your ability to edit depends more on your familiarity with the NLE than the software itself, and as it’s a pretty standardized process, you won’t find major differences between programs. With both Premiere & Resolve, I was able to cut together a multicam within an hour or so of messing around; no need to open up a manual. But if you go beyond basic edits, Premiere shows its pedigree; its tools are more refined and offer more options for power users.



One of my biggest problems with Premiere is that in order to use GPU acceleration, you have to use linear color. For some (fundamental, intractable) reason, this means your crossfades look terrible. I’ve tried everything from custom presets to manually automating opacity, to no avail; if you want smooth crossfades in Premiere, you need to turn off the GPU. For the convenience of not having to switch settings back and forth to preview crossfades, I’m giving this to Resolve.



Resolve just added a full mixer, allowing you to use it as a DAW as well as an NLE. It would get my vote for that, but it doesn’t affect me either way; I use a set of dedicated tools for mixing, and I won’t be changing that anytime soon.



One of the requirements for editing multi-camera shoots is being able to synchronize clips. I’m sorry to say it, but as of this post, audio sync in Resolve was not useable for me. I tested it with multiple concert shoots, all with live audio on the video tracks. While Premiere had no problems aligning multiple audio & video clips accurately, Resolve didn’t put the clips anywhere close to where they should have been (we’re talking mismatches of 20 minutes). I’m sure it works fine with timecode, but I’m not going to drop $2,000 on a Lockit kit when audio sync should work just as well.


Color Correction & Grading


This shouldn’t surprise anyone; Resolve started out as a professional grading application and it’s still centered around its roots. Despite being a novice at grading (and having only used the program for a day), I was able to get my images looking a lot better a lot faster with Resolve. For grading multi-cam shoots, node groups are a godsend. I haven’t even begun to delve into qualifiers and trackers, but they’re insanely powerful tools.

Plugins & Effects


Despite being a relatively young program, Resolve has an impressive suite of effects. Its core effects (crossfades, titles) are a little more accessible than Premiere’s, and it includes some very capable noise reduction and film grain algorithms (in Premiere, you’d need to find or pay for 3rd-party plugins for that functionality).

However, Premiere’s effects have a lot more parameters and options, and their text generator is much more developed. Also, you can keyframe parameters directly within the editor; in Resolve, that process is a little clunky and involves going back and forth between separate windows. I’m calling a tie.

Render Speed


Just like with playback, I have to turn off GPU acceleration in Premiere to get smooth dissolves, which takes away a lot of speed. With GPU acceleration in Resolve, renders are very fast; about 1x realtime even with grading, effects and compositing applied. That’s very impressive.


Render Quality


This is one of the areas where Premiere really demonstrates its maturity; Premiere has a wide range of codecs and compression options, and it does a great job of rendering high-quality video without creating huge files. By comparison, Resolve is lagging pretty far behind. Render options are limited and confusingly arranged, and the quality of the files that the presets produce is noticeably worse. I was only able to get good results by rendering out to visually lossless codecs like DNxHR HQX, which are intermediate rather than delivery formats.


The workaround? Render our the grade in lossless formats, and do the final pass in Adobe Media Encoder. This isn’t a perfect solution, but it’s my best option for now.



I’m only using the free version right now, but $300 for lifetime upgrades is a great value. I’m paying $50/mo. for Creative Cloud, and while I use several other programs in the suite, after a couple years it’ll have surpassed the price of Resolve.

The Verdict

Both programs are very powerful, and you couldn’t go wrong with either; I know I’ll be using both for years to come. A few more versions down the road, I may end up using Resolve as my NLE, but right now it’s not quite up to the job. In the end, I ended up sticking with both for their original strengths (surprise); Resolve for grading, and Premiere for editing.

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Digital Tape, Pt. 4: Smoothing Out Tone


The analog vs. digital debate is a topic unto itself, and I don’t have any interest in engaging that; I use analog tools where appropriate, and digital tools for the same reason. However, a little backstory is appropriate. When you’re recording entirely in the analog domain, every action you take loses you some high frequencies. Run it through a desk? Lose some HF. Run it through a compressor? Lose some HF. Record to tape? Bounce to tape? Play the tape? Lose some HF. Along the way, we got used to the tone that came from all those analog passes. Then along came digital, promising superior fidelity with no signal loss. Bad converters aside, the consensus? It’s harsh! Cold! Brittle! Digital, with all its superior fidelity, often seems to be missing a little charm.

These days we have the best of both worlds; if you want the sound of tape, tubes or transformers, you can record through tape, tubes and transformers without having to worry about signal loss down the road. I didn’t grow up on vinyl, and I don’t have any nostalgia for low fidelity; however, there is something to the way analog gear handles high frequencies. I’m going to explore a couple of options for emulating that response with more precision.


For tonal content, DMG Essence is an absolutely amazing tool. While it’s billed as a mastering de-esser, it’s really a sub-band processor that allows you to selectively compress specific frequencies. In practice; let’s say you’ve got a vocal track that sounds fine when the singer’s quiet, but sounds harsh when they hit the chorus. You can isolate the offending frequency and set the threshold so it only engages at high levels, effectively smoothing out the signal. This technique can be used with high ratios to deal with resonances in instruments or amps (even feedback), but if used subtly (with low ratios & gentle filter slopes) it can emulate the way tape attenuates high frequencies when pushed. With experimentation you can achieve this effect very transparently, without the inherent noise or distortion of tape.


OD DeEdger offers a similar feature set that’s tailored to transient content. While the controls are much simpler, the concept is the same; pick a frequency, pick a Q, and set your threshold. DeEdger is perfectly suited to taming grating percussion, crunchy amps, and the high-mid sharpness you can get from cheaper condensers. While I haven’t tried them myself, oeksound’s Soothe and Spiff seem to be well-suited to taming harshness as well, and they’re creating some good buzz in audio circles.

There’s no denying that analog gear is a lot more fun to play with (and makes for a more impressive studio shot), but not owning a $7,000 preamp is no excuse for not having euphonic tone. Hope this series helped you get a little closer to that!

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Digital Tape, Pt. 3: Gain-Staging Drums

Note: the following post makes use of Klanghelm VUMT and TDR Limiter 6. To get the most out of these posts, head over to their sites and support some awesome developers!


In our last post we covered how to apply analog gain-staging practices to melodic material. However, our ears respond much differently to percussive material, which is composed of transients instead of tones.


A PPM meter responds much more quickly to level changes, and as such it’s better suited to drums and percussion. Using VUMT, pull up a simple DIN meter and apply it to a snare or kick track. Adjust the pre-fader clip volume so that the needle is reading a little under 0.

Here’s an example of a snare track after I’ve adjusted the gain:


You’ll notice that while the hits are fairly consistent, there are couple hits around the 1/3rd mark that are much louder than the rest. In our last post, we dealt with this problem by manually adjusting the volume; while this is a completely valid approach, it’s not always an option to go through hundreds of hits by hand. Instead, let’s take an analog approach.

Back in the days of tape, meters weren’t fast enough to respond to “peaks” as we think of them today; when your meters are sitting around 0PPM, loud hits like the ones above can easily slip through without being caught. However, tape responds much differently than a digital recorder. When audio reaches maximum level, tape goes into saturation, clipping peaks while generating harmonic distortion. While this effect is rarely desired for melodic material, for drums it not only preserves most of the dynamic impact, but can actually enhance the sound. In the digital domain, we can selectively use this effect to enhance our drums in a much more transparent way.

clipperTDR Limiter 6 ($60 USD) is an incredible plugin that’s perfectly suited to the task. We’re going to use just the clipper module here, set to a drive of 0dB, threshold of -6dB, knee of 4dB, and with the mode set to “open.” This gives what’s called a “soft clip” that’s very similar to the clipping properties of tape. If your peaks are sitting around 0PPM, the clipper shouldn’t be engaging regularly; at most, it should be shaving off 0.5 dB from your hits. But on those few errant peaks, the clipper will dig in and clip off the tops, just like tape. You shouldn’t be clipping more than 3dB unless you want audible distortion, and there are better tools for that job than a clipper. What we’re doing is emulating a property of tape to transparently control our peaks. While Limiter 6 is a great option, any plugin with a soft clipper should be able to accomplish the task.

Just as before, we don’t need to delve into the technical side of this to go on with our mix; by following the steps above, you’ll have drum hits that are dynamic and consistent in level with the rest of your tracks.

No matter what methods you use, I can’t stress enough how important it is to take the time to gain-stage your tracks correctly before you begin your mix. By properly preparing your audio, you set yourself up for a fast, intuitive mix session that puts the best of both worlds into practice.

As an addendum, there are a few other techniques I use to selectively emulate some of the best properties of tape. Read on…

>> Digital Tape, Pt. 4: Getting Smooth Tones

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Digital Tape, Pt. 2: Gain-Staging Vocals

Note: the following post makes use of Klanghelm VUMT, an incredible plugin that costs about $15 USD. To get the most out of these posts, head over to the site and support an awesome developer!


Hugh Robjohns begins his article on VU & PPM meters with the following quote:

These are both, strictly speaking, obsolete analogue metering formats! In short, the VU meter shows an averaged signal level and gives an impression of perceived loudness, while a PPM indicates something closer to the peak amplitude of the input signal. However, in our modern digital world, neither meter really performs adequately.

He has a point; if you’re mastering for broadcast, you want a digital meter that’s able to respond with pinpoint accuracy. However, I’d like to focus on part of that quote:

The VU meter … gives an impression of perceived loudness.

While digital meters do a great job of letting you know if your signal is clipping, they don’t correlate much with what you’re actually hearing. One of the reasons people love mixing outside the box is that analog meters move the way the music does. So let’s set up some analog meters.


Klanghelm VUMT is my favorite meter plugin, and it has far more features than we’ll need today. For now, just bring it up as a simple VU meter and apply it to a vocal track. Adjust the track volume (pre-fader) in the DAW so that the meter is reading around 0VU for sustained vocal phrases.

After you’ve set your gain, you’ll notice that there are some big variations in level; some phrases will be close to clipping, while others will barely move the needle. We’re going to take another page from the analog book and do some level-riding. If you were mixing on an analog desk, this would mean keeping your finger on the fader and manually moving it up and down to compensate for level changes. In the digital world, we can use our DAW to adjust the gain. In my case, I’ll use REAPER to cut it up into clips and adjust their gain, but you could also do this with trim or pre-fader volume envelopes.

Here I’ve got a vocal track with a lot of dynamic range. It was a live recording, so on top of the musical dynamics the singer was changing his distance to the mic, creating a wide range of levels:


First I’ll cut the track into sections (verse, chorus) and balance the gain between them. Then I’ll go in with more of a fine-toothed comb, adjusting individual phrases and words:


The goal isn’t to kill the dynamics of the track, and you should make sure to audition each gain change to make sure it sounds natural. Here are the two clips compared, with the original on top:


As you can see, the levels on the bottom clip are much more consistent, while still retaining plenty of dynamics; the meter should be reading 0VU more consistently as well.

Now, go through the same process with any other tracks that contain melodic material; in other words, anything but drums. Your lead vocal is your loudness reference, so use your ears while you’re matching levels; however, you can aim for 0VU for legato tones (vocals, lead guitar, bass, horns, and bowed instruments), while more percussive material (rythmic guitar strums, hard piano chords) will clock in closer to -7/-5VU. Again, use your DAW to ride the levels and even out big variations.

You’re welcome to dive into the technical aspects of all this, like how to convert VU to dBFS and how headroom applies to the digital domain. But regardless of that, if you follow the steps above then you’ll meet all the goals we set in the previous post; consistent levels, adequate headroom, and good signal-to-noise. Most importantly, when your faders are at unity your tracks will all sound equally loud. With this foundation, the mix process becomes much more intuitive.

I’ve used the term “digital tape” because this process emulates how you’d gain-stage if you were recording to an analog medium. However, in the next post we’ll be looking into actually emulating some properties of tape in the digital domain, and how to use PPM meters to gain-stage percussive content. Read on…

>> Digital Tape, Pt. 3: Gain-Staging Drums

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Digital Tape, Pt. 1: Analog Gain-Staging in a Digital World


In my opinion, gain-staging is the most important step of the mix process. Properly preparing your audio sets the foundation for your mix; unfortunately, this step is often neglected or misunderstood. I’m starting a series of short articles on how to apply analog mixing practices to the digital domain, in the hopes of simplifying this fundamental process.

One of the biggest differences between analog and digital is the way the equipment responds to level. When recording through tape & tube equipment, there’s a sweet spot in the middle of the range where fidelity and signal-to-noise is optimized. Get the level too low and your signal gets lost in hiss; get the level too high and it’ll distort. 24-bit digital is an entirely different picture; get the level too low and it’s not much of a problem, as long as your peaks aren’t clipping. Compounding this problem is the fact that digital meters don’t do the best job of representing what we actually hear.

Because we’re not fighting for a smaller sweet spot, we tend to be more relaxed when it comes to gain-staging in the digital world. This is fine for recording, but it creates big problems when you enter the mix stage, because:

  • if your levels aren’t consistent, your faders won’t be representative of the actual mix.
  • if you have multiple tracks that are close to peaking, they’ll clip your master bus when combined.
  • saturation, distortion, and analog-emulation plugins are level-dependent just like analog gear.

What we want is tracks whose levels are consistent in relation to each other, high enough to optimize signal-to-noise, low enough to leave plenty of headroom on the master bus, and gain-staged for use with analog emulations. How do we do this? Read on…

>> Digital Tape, Pt. 2: Gain-Staging Vocals

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How to Mix for $500


I’m a firm believer in using a small set of tools that you know inside & out, and this applies very strongly to my audio workflow. Rather than downloading massive plugin bundles, I’d highly recommend learning to work with just one DAW, one EQ, one compressor, and one reverb until you’re intimately familiar with the principles of each. 90% of my work is done with 4 very powerful pieces of software that I’ve listed below, and I’d highly recommend them to anyone who’s doing audio work at any level.

DAW // REAPER ($60)

Your DAW is your workflow, and workflow is the most important part of the mix process. I’ve been using REAPER for years, and I wholeheartedly recommend it for everything from editing podcasts & recording live shows to mixing & mastering studio albums. REAPER isn’t just cheaper than Pro Tools & Logic; in my opinion, it’s better. There are so many things I love about this DAW; free updates for life, incredible stability, fast & intuitive clip editing tools, flexible audio routing, great hardware integration, and perfect audio quality. I was able to use it competently after a few days of experimentation, but 2 years later I bought a hard copy of the manual and realized how many incredible features I hadn’t even discovered.

EQ // DMG EQuilibrium ($230)

EQuilibrium is the most expensive software on this list, but there are ways around that (see below). Even at sticker price, I’d recommend this EQ to anyone who’s serious about mixing audio. Since its release in 2013, it’s maintained its place as the last EQ you’ll ever need to buy. It’s held its own in listening tests against $6,000 analog EQs, while a vast selection of EQ curves, a wide range of modes (analog, digital, linear-phase), and an extreme attention to audio quality set this EQ apart from the rest.

Compressor // Klanghelm DC8C ($25)

DC8C is deceptively inexpensive for what it is; one of the best-sounding software compressors on the market. It’s absolutely incredible on guitar, bass, and drums, and its character settings allow you to dial in your sound in seconds. While you can use its streamlined GUI for fast results; a deep feature set makes this the perfect tool to learn the ins & outs of dynamics processing.

Compressor // TDR Kotelnikov GE ($50)

Kotelnikov is another plugin that punches above its price point; it’s one of the cleanest compressors I’ve heard. I’m a big fan of transparent compression, and Kotelnikov is one of the only compressors that I can throw on a vocal track, barely tweak the settings, and have it invisibly control the dynamics (it’s equally great on the master bus). As with EQuilibrium, meticulous attention has been paid to audio quality; this is good DSP.

Limiter // TDR Limiter 6 GE ($60)

Another incredible offering from Tokyo Dawn; Limiter 6 is a full master bus in a single plugin, combining a compressor, clipper, and 3 limiting stages with a comprehensive loudness analyzer. Whether you’re mastering full albums or just bringing your demos up to commercial volume, Limiter 6 can be as simple or as complex as you need. I’d say it’s worth it for the clipper and EBU meter alone.

Reverb // Exponential Audio PhoenixVerb ($120)

PhoenixVerb has recently been superseded by NIMBUS, but for half the price you can get 90% of the features and sound of EA’s flagship reverb. PhoenixVerb is known for a smooth, natural sound that’s perfectly suited to acoustic genres of music, and for giving dry/isolated tracks a real sense of space & position. It’s a reverb that doesn’t demand attention, but when you turn it off it feels like the mix loses a whole spatial dimension. I’ve gone through some very high-quality reverbs before settling on PhoenixVerb, and now it’s pretty much all I use.

One thing that really changed the game for me was realizing that you can buy software secondhand. Used software is new software, but you can often find all the plugins I’ve listed above for half their sticker price. I wrote a quick post on where to look for software online, so check that out here.

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How to Buy Used Software

It took me a couple years to realize this, but you can buy audio software secondhand. Used software is the same as new software (for obvious reasons), but with a little patience you can buy your plugins for far less than their sticker price (usually about 50% off). Here’s a list of places where you can buy your licenses; all of these are reputable communities with checks & balances to make sure you’re not paying for pirated software.

As always, make sure to pay via PayPal. I’ve been buying from these forums for a few years now, and I’ve only had good experiences!

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Bessel Curves: The Invisible Filter


Around a year ago I came across a very enlightening post by Bob Macc on the sonic consequences of treating your tracks with aggressive filters. You should go check it out yourself, but the big idea is that using steep filters causes significant phase distortion that reaches all the way up and down the frequency spectrum. Phase distortion removes clarity and impact from your mix, so in short, it’s not good. With that in mind, I’ve come up with some good practice points for filtering your audio.

First, use gentle filter slopes. 6db/octave is my go-to for LPFs, and 12db/octave is my go-to for HPFs. This lessens the impact of phase distortion, but most importantly, it sounds more natural and there’s less risk of accidentally cutting out some good stuff.

Second, use a Bessel curve. Bessel filters are linear phase, which means there’s near-zero phase distortion when they’re implemented correctly. While some EQs have linear phase DSP options, remember that if you’re using steep filter slopes, linear phase will cause distortion too; it’s the combination of gentle slopes and Bessel curves that’ll preserve your audio.

This advice is in line with the Hippocratic philosophy of mixing, e.g. “do no harm.” I’m mixing mostly acoustic music, hut it doesn’t matter if you’re mixing orchestral, folk, punk, or EDM; phase distortion from filters is rarely something you want. Following the two points above will save you a lot of mix clarity and let you put the distortion where you want it.

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Recording is Subtractive


I want to take on the mindset that recording is additive.

Here’s what I mean; when musicians are shopping around for a studio, it’s common for the engineer to mention that such-and-such gear was used by such-and-such artist. They do this because it works; people love to hear it. There’s a mythology around famous gear that usually sounds like this: If you sing through Sinatra’s mic, you’ll sound like Sinatra. If you’re mixing through a Neve, you’ll sound like Steely Dan. It might not be explicit, but it’s often implied that the sound comes from the gear. Selling a studio based on its gear is a problem unto itself, but this all stems from the idea that recording is additive; that every piece of gear you record through is going to add something to the sound until your music comes out the other end sounding like a finished product.

But when you look at it from a technical perspective, it’s actually the opposite. The microphone, the first item in the signal chain, is essentially a filter. It’s rolling off the frequencies at both ends of the audible spectrum, and adding resonance and noise. The next item is the microphone preamp; a “clean” preamp is going to do its best to preserve all the information in the original signal, while a “colored” preamp is going to lose some information, usually transients & high frequencies. If you’re tracking through a compressor, you’re decreasing dynamic range. If you’re recording to tape, you’re adding noise, losing more transient impact, and further reducing the frequency range. If you’re recording to digital, you’ll need high-quality conversion to translate the analog signal without degradation.

When you look at recording through this lens, you can see that at each stage you’re losing information. Each successive stage adds noise and distortion while reducing the spectrum of volume and frequency. Each piece of gear is subtractive.

Of course, that’s not a bad thing. Think of a photograph; a sharp, high-resolution image of your face will probably expose too much detail. A little blur and distortion can be attractive, and that principle goes for audio too. But imagine if someone told you that since their camera was used to take photos of models, you’ll look like a model through that camera too. We have an intuitive understanding of the visual medium, so we wouldn’t fall for something like that; we know a camera can only capture the image, not create it. It’s the same way with audio.

So here’s the basic idea: from microphones & preamps to tape & digital, each piece of gear can only capture what’s there to begin with, and it can only remove information from that point. For engineers & studio owners, that means we need to stop treating gear like it’s magic; the selling point should be our results, not our tools. For artists, that means you can’t rely on the gear to give you your sound. Start with the best performance you can give, on the best instrument you can find, in the best room you can rent, and it won’t get any better than that.

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